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  • aac14973aa (Auto)update libjingle 71116846-> 71117224 buildbot@webrtc.org 2014-07-14 20:22:21 +00:00
  • 5be649fcfc Add a facility to the Thread class to catch blocking regressions. tommi@webrtc.org 2014-07-14 20:21:36 +00:00
  • 242068d58c A step towards changing StatsReport::Value::name to an enum. The stats reporting code does a lot of unnecessary string copying. This is a step in the direction of removing that and forcing use of only known constants. tommi@webrtc.org 2014-07-14 20:19:56 +00:00
  • 03505bcb7a Make StatsCollector depend on always having a valid session pointer. This is required since the session pointer is currently used on multiple threads but there's no synchronization code to guard it. I'm removing the set_session() method and session() getter since they would cause problems if used without synchronization. tommi@webrtc.org 2014-07-14 20:15:26 +00:00
  • b5348c64bb Minor refactoring of the session classes. Make member variables that never change and are touched on multiple threads, const. Move implementations of setters/getters of variables that can change, into the cc file in preparation of adding thread correctness checks. tommi@webrtc.org 2014-07-14 20:11:49 +00:00
  • d8524348bb (Auto)update libjingle 71107853-> 71115715 buildbot@webrtc.org 2014-07-14 20:05:09 +00:00
  • b92f6f9371 (Auto)update libjingle 71099685-> 71107853 buildbot@webrtc.org 2014-07-14 18:22:37 +00:00
  • a4da771914 Fix deadlock in Android stopCapture() call. glaznev@webrtc.org 2014-07-14 17:01:53 +00:00
  • 5f43ce6784 Fix a type cast issue for compiling webrtc with BoringSSL. jiayl@webrtc.org 2014-07-14 16:42:46 +00:00
  • e04cb0eb81 (Auto)update libjingle 70948025-> 70959275 buildbot@webrtc.org 2014-07-14 14:54:16 +00:00
  • 9bef551ba1 GN: Fix include paths for WebRTC in Chromium build. kjellander@webrtc.org 2014-07-13 09:02:54 +00:00
  • 9e1acc8728 Fix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 . tommi@webrtc.org 2014-07-11 20:33:39 +00:00
  • dd6780d85d Remove always-true expression. tommi@webrtc.org 2014-07-11 19:34:54 +00:00
  • eec6ecdb1e Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC. --- tommi@webrtc.org 2014-07-11 19:09:59 +00:00
  • 180e516bef Thread annotate RTCPSender. pbos@webrtc.org 2014-07-11 15:36:26 +00:00
  • 336e8e8f50 Fixing memcheck leak suppressions for XMPPClient tests. pbos@webrtc.org 2014-07-11 13:44:45 +00:00
  • 168f23faa5 Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. stefan@webrtc.org 2014-07-11 13:44:02 +00:00
  • ccbed3b3c4 Implement unittest SetRecvCodecsAcceptDefaultCodecs. pbos@webrtc.org 2014-07-11 13:02:54 +00:00
  • a1bfcad3a3 Cast payload types to int for logging. pbos@webrtc.org 2014-07-11 12:33:45 +00:00
  • fb2e7c22a0 Document that channels are stored contiguously in AudioBuffer aluebs@webrtc.org 2014-07-11 11:40:48 +00:00
  • d212ffcfc6 Remove unnecessary build message. tommi@webrtc.org 2014-07-11 11:15:35 +00:00
  • 4ef438e2de Remove the send-side cname getter APIs from voice and video engine. stefan@webrtc.org 2014-07-11 09:55:30 +00:00
  • 0f426685e1 Roll chromium_revision 280876:282462 henrikg@webrtc.org 2014-07-11 08:10:19 +00:00
  • cb973686e8 roll libyuv to r1033 for clang-cl support on windows. BUG=chromium:391927 TESTED=manual testing libyuv compiles with clang-cl R=andrew@webrtc.org fbarchard@google.com 2014-07-10 23:40:15 +00:00
  • b614d0626f Rebase webrtc/base with r6655 version of talk/base: cls to port: r6633,r6639 (there is no cl in between that affects base and all other talk/base cls took care of webrtc/base as well (see r6569, r6624)): svn diff -r 6632:6639 http://webrtc.googlecode.com/svn/trunk/talk/base > 6655.diff sed -i.bak "s/talk_base/rtc/g" 6655.diff patch -p0 -i 6555.diff henrike@webrtc.org 2014-07-10 22:47:02 +00:00
  • 72491b9a90 Count total bytes sent in RTPSender::Bytes(). pbos@webrtc.org 2014-07-10 16:24:54 +00:00
  • 0422100818 Fix data race in VCMTiming::ResetDecodeTime. pbos@webrtc.org 2014-07-10 15:25:37 +00:00
  • bd9c0920ec Skip encoding in fake VP8 encoder. pbos@webrtc.org 2014-07-10 13:21:40 +00:00
  • 7ae9108b60 Remove more unused tsan suppressions and fix call test passing the same decoder to multiple received streams. andresp@webrtc.org 2014-07-10 10:35:12 +00:00
  • 91f1752f2d Support VP8 encoder settings in VideoSendStream. pbos@webrtc.org 2014-07-10 10:13:37 +00:00
  • 8f1512140e Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel. andresp@webrtc.org 2014-07-10 09:39:23 +00:00
  • 5bde66e913 audio_processing: Updates aec_core_sse2.c with changes made to aec_common.h bjornv@webrtc.org 2014-07-10 08:09:50 +00:00
  • 555fc78f27 Neon version of SubbandCoherence() bjornv@webrtc.org 2014-07-10 08:03:11 +00:00
  • ac800c8004 Neon version of rftbsub_128() bjornv@webrtc.org 2014-07-10 07:53:13 +00:00
  • 5ac876bae0 Revert "Remove remains of WEBRTC_NO_STL." (rev 6641). Reason breaks linux_memcheck. andresp@webrtc.org 2014-07-10 07:41:59 +00:00
  • e91ba268e3 Revert 6643 "Revert 6637 "Revert 6636 "Roll chromium_revision 28..." henrikg@webrtc.org 2014-07-10 07:08:32 +00:00
  • 02dce515d3 Revert 6637 "Revert 6636 "Roll chromium_revision 280876:281479"" henrikg@webrtc.org 2014-07-10 06:56:45 +00:00
  • 72670206db (Auto)update libjingle 70813271-> 70818369 buildbot@webrtc.org 2014-07-09 20:40:58 +00:00
  • 47d1c98a4e Remove remains of WEBRTC_NO_STL. andresp@webrtc.org 2014-07-09 20:18:28 +00:00
  • 10ef8fe611 Create FullScreenChromeWindowDetector in DesktopConfigurationOptions::CreateDefault. jiayl@webrtc.org 2014-07-09 19:41:32 +00:00
  • 4b1f330b4f Fix a bug in SocketAddress where "a.b.c.d:1" and "b.b.c.d:1" are incorrectly considered equal. jiayl@webrtc.org 2014-07-09 19:14:24 +00:00
  • 7af12be781 Thread annotations for vie_encoder.cc/.h stefan@webrtc.org 2014-07-09 14:46:31 +00:00
  • e7771d07c8 Revert 6636 "Roll chromium_revision 280876:281479" henrikg@webrtc.org 2014-07-09 13:15:50 +00:00
  • 543da997f2 Roll chromium_revision 280876:281479 henrikg@webrtc.org 2014-07-09 13:03:39 +00:00
  • 045a9b17da Remove unnecessary race suppressions copied from chromium. And added suppressions to allow to run tests with gtest_parallel in which case some new races were showing up. andresp@webrtc.org 2014-07-09 11:44:34 +00:00
  • b8e9e44eac Add full stack test cases with a fake network pipe. stefan@webrtc.org 2014-07-09 11:29:06 +00:00
  • e9cefdef68 Improve libjingle's ASSERT and VERIFY macros on Windows. This change has the effect that when using a debugger, a failing ASSERT/VERIFY will break exactly where the failing expression is and not two callstacks up. Minidumps (for debug builds) will also have the failing expression at the top of the call stack. tommi@webrtc.org 2014-07-09 08:04:12 +00:00
  • 01bda2068b Fixed the stats problem when new track is using the same ssrc as the previous track. xians@webrtc.org 2014-07-09 07:38:38 +00:00
  • b753762ce6 delay_estimator: Increases test coverage and makes input spectrum const bjornv@webrtc.org 2014-07-09 06:40:09 +00:00
  • 12b4efefdd Implement a work around for Chrome full-screen tab switch on Mac. jiayl@webrtc.org 2014-07-08 22:05:24 +00:00
  • e55641d4f7 Neon version of rftfsub_128() bjornv@webrtc.org 2014-07-08 21:12:23 +00:00
  • 55535d4e58 (Auto)update libjingle 70711261-> 70733822 buildbot@webrtc.org 2014-07-08 18:18:55 +00:00
  • d11bec40b2 Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel. andresp@webrtc.org 2014-07-08 14:32:58 +00:00
  • 3d7da88e06 Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable. stefan@webrtc.org 2014-07-08 13:59:46 +00:00
  • ecb8723402 Change Timing::WallTimeNow to be static. tommi@webrtc.org 2014-07-08 12:48:29 +00:00
  • 62bafae661 Some refactoring inside rtp_rtcp/. pbos@webrtc.org 2014-07-08 12:10:51 +00:00
  • 241a9b0b65 Fixing compile error. phoglund@webrtc.org 2014-07-08 11:48:37 +00:00
  • 22292df53b Adding explicit check for using dummy file devices. phoglund@webrtc.org 2014-07-08 11:39:19 +00:00
  • 33d110d8ea Tight data race suppressions around thread_posix. andresp@webrtc.org 2014-07-08 10:36:39 +00:00
  • af38f4e511 Extract RTP-header SSRC inline in Call. pbos@webrtc.org 2014-07-08 07:38:12 +00:00
  • a70be68f65 Disabling shared socket mode for TURN ports. This is done as currently when TURN server also used as STUN server, binding responses will be handed over to TURN port, which simply discard these messages, as requests are originated from StunPort. mallinath@webrtc.org 2014-07-07 20:47:24 +00:00
  • 3c637cdaa5 Clean data races from system_wrappers_unittests. andresp@webrtc.org 2014-07-07 20:37:39 +00:00
  • 285e9bc84d Fix potential deadlock in webrtc/system_wrappers/source/logging_unittest.cc. andresp@webrtc.org 2014-07-07 20:27:33 +00:00
  • 5f2c81c17f webrtc/base: Fixes miss in base.gyp for windows. See https://code.google.com/p/webrtc/source/browse/trunk/talk/libjingle.gyp?r=6503#764 for the corresponding condition. henrike@webrtc.org 2014-07-07 17:42:45 +00:00
  • ba93f9a986 drmemory flaky: EndToEndTest.RestartingSendStreamPreservesRtpState[WithRtx] suppressed on drMemory. henrike@webrtc.org 2014-07-07 16:52:19 +00:00
  • 161f808500 Add test for VideoEncoder setup/teardown. pbos@webrtc.org 2014-07-07 14:22:35 +00:00
  • 2bb1bdab8d Preserve RTP states for restarted VideoSendStreams. pbos@webrtc.org 2014-07-07 13:06:48 +00:00
  • 73823cafa4 Add initial gn build files for video_coding and video_processing. stefan@webrtc.org 2014-07-07 11:46:43 +00:00
  • 03c817e405 Fix pacer to accept duplicate sequence numbers on different SSRCs. pbos@webrtc.org 2014-07-07 10:20:35 +00:00
  • b941fe8098 Fix data races related with traces in bitrate estimator test. andresp@webrtc.org 2014-07-07 08:50:48 +00:00
  • bd249bc711 Remove GetDefaultConfigs() from Call. pbos@webrtc.org 2014-07-07 04:45:15 +00:00
  • 7832648824 Add missing break introduced in r6603. stefan@webrtc.org 2014-07-04 17:04:00 +00:00
  • bee164a214 Fix test issues and a win compile error introduced with r6605. stefan@webrtc.org 2014-07-04 17:00:06 +00:00
  • 875ad49dee Revert conversion from TickTime to int64_t in paced sender. stefan@webrtc.org 2014-07-04 15:27:55 +00:00
  • 8faa5db992 Add pbos@webrtc.org as owner for webrtc/test/. pbos@webrtc.org 2014-07-04 13:49:17 +00:00
  • b9f5453e29 Add boilerplate code for H.264. stefan@webrtc.org 2014-07-04 12:42:07 +00:00
  • d8440f7c45 Have Opus follow Chromium revisions tina.legrand@webrtc.org 2014-07-04 12:13:08 +00:00
  • 20c1f56992 Configure RTX send status on new modules. pbos@webrtc.org 2014-07-04 10:58:12 +00:00
  • 88e0dda475 Introduces PacedVideoSender to test framework and moves the Pacer to use Clock. stefan@webrtc.org 2014-07-04 09:20:42 +00:00
  • 614000d638 Adding pbos as video/ owner and removing persons never working with this folder. mflodman@webrtc.org 2014-07-04 08:50:14 +00:00
  • c5e53dde71 Revert 6597 "Roll chromium_revision 280876:281094" kjellander@webrtc.org 2014-07-04 07:53:46 +00:00
  • cb1df98093 Roll chromium_revision 280876:281094 kjellander@webrtc.org 2014-07-04 07:31:19 +00:00
  • 720964faac Fix memcheck error in r6594. marpan@webrtc.org 2014-07-03 21:14:07 +00:00
  • 11bea8977e GN: Implement BUILD.gn for common_video. kjellander@webrtc.org 2014-07-03 17:04:12 +00:00
  • c8364539d3 Fix for FEC decoding with sequence number wrap-around. marpan@webrtc.org 2014-07-03 16:49:30 +00:00
  • 69ef9911e4 delay_estimator: Allows dynamically used history sizes bjornv@webrtc.org 2014-07-03 14:59:03 +00:00
  • 224a140339 Make experimental NS API not purely virtual aluebs@webrtc.org 2014-07-03 13:41:39 +00:00
  • c0ba4392f1 common_audio: Removes macro WEBRTC_SPL_SHIFT_W16 bjornv@webrtc.org 2014-07-03 13:38:53 +00:00
  • 38214d53db EchoCancellationImpl::ProcessRenderAudio: Use float samples directly kwiberg@webrtc.org 2014-07-03 09:47:33 +00:00
  • a82f9a243d Add Tsan2 to .gitignore andresp@webrtc.org 2014-07-03 08:23:58 +00:00
  • dfdaeb92d8 Removed old code and default implementations. asapersson@webrtc.org 2014-07-03 07:35:21 +00:00
  • 9c89e932c9 WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering. (Including another fix here, https://review.webrtc.org/16779004/, to make the test run) braveyao@webrtc.org 2014-07-03 05:59:22 +00:00
  • 3ffa1f917e (Auto)update libjingle 70422491-> 70424781 buildbot@webrtc.org 2014-07-02 19:51:26 +00:00
  • b25b08b302 Remove tools/resources kjellander@webrtc.org 2014-07-02 15:48:29 +00:00
  • 93426cd2ff Implement BUILD.gn for desktop_capture. jiayl@webrtc.org 2014-07-02 15:47:12 +00:00
  • 33586c83b1 Make deadlock suppressions less generic. andresp@webrtc.org 2014-07-02 14:19:05 +00:00
  • 1295dc6a23 Possibly fix deadlock happening due to unregister/register modules as switching between AST and TSO estimators. andresp@webrtc.org 2014-07-02 13:23:19 +00:00
  • 0bb9fac98c (Auto)update libjingle 70343444-> 70394475 buildbot@webrtc.org 2014-07-02 11:54:09 +00:00
  • 895698067c Roll chromium 280149:280876. marpan@webrtc.org 2014-07-02 02:50:05 +00:00
  • d8a9069080 (Auto)update libjingle 70340027-> 70343444 buildbot@webrtc.org 2014-07-01 19:26:43 +00:00